Spectral translation/ folding in the subband domain

ABSTRACT

The present invention relates to a new method and apparatus for improvement of High Frequency Reconstruction (HFR) techniques using frequency translation or folding or a combination thereof. The proposed invention is applicable to audio source coding systems, and offers significantly reduced computational complexity. This is accomplished by means of frequency translation or folding in the subband domain, preferably integrated with spectral envelope adjustment in the same domain. The concept of dissonance guard-band filtering is further presented. The proposed invention offers a low-complexity, intermediate quality HFR method useful in speech and natural audio coding applications.

[0001] The present invention relates to a new method and apparatus forimprovement of High Frequency Reconstruction (HFR) techniques,applicable to audio source coding systems. Significantly reducedcomputational complexity is achieved using the new method. This isaccomplished by means of frequency translation or folding in the subbanddomain, preferably integrated with the spectral envelope adjustmentprocess. The invention also improves the perceptual audio qualitythrough the concept of dissonance guard-band filtering. The proposedinvention offers a low-complexity, intermediate quality HFR method andrelates to the PCT patent Spectral Band Replication (SBR) [WO 98/57436].

[0002] Schemes where the original audio information above a certainfrequency is replaced by gaussian noise or manipulated lowbandinformation are collectively referred to as High FrequencyReconstruction (OR) methods. Prior-art HFR methods are, apart from noiseinsertion or non-linearities such as rectification, generally utilizingso-called copy-up techniques for generation of the highband signs Thesetechniques mainly employ broadband linear frequency shifts, i.e.translations, or frequency inverted linear shifts, i.e. foldings. Theprior-art HFR methods have primarily been intended for the improvementof speech codec performance. Recent developments in highbandregeneration using perceptually accurate methods, have however made HFRmethods successfullly applicable also to natal audio codecs, codingmusic or other complex programme material PCT patent [WO 98/57436].Under certain conditions, simple copy-up techniques have shown to beadequate when coding complex programme material as well These techniqueshave shown to produce reasonable results for intermediate qualityapplications and in particular for codec implementations where there aresevere constraints for the computational complexity of the overallsystem.

[0003] The human voice and most musical instruments generatequasistationary tonal signals that emerge from oscillating systems.According to Fourier theory, any periodic signal may be expressed as asum of sinusoids with frequenciesƒ, 2ƒ, 3ƒ, 4ƒ, 5ƒ etc. where ƒ is thefundamental frequency. The expressed as a sum of sinusoids withfrequencies ƒ, 2ƒ; 3ƒ; 4ƒ, 5ƒ etc. where ƒ is the fundamental frequency.The frequencies form a harmonic series. Tonal affinity refers to therelations between the perceived tones or harmonics. In natural soundreproduction such tonal affinity is controlled and given by thedifferent type of voice or instrument used. The general idea with HFRtechniques is to replace the original high frequency information withinformation created from the available lowband and subsequently applyspectral envelope adjustment to this information. Prior-art HFR methodscreate highband signals where tonal affinity often is uncontrolled andimpaired. The methods generate non-harmonic frequency components whichcause perceptual artifacts when applied to complex programme material.Such artifacts are referred to in the coding literature as “rough”sounding and are perceived by the listener as distortion.

[0004] Sensory dissonance (roughness), as opposed to consonance(pleasantness), appears when nearby tones or partials interfere.Dissonance theory has been explained by different researchers, amongstothers Plomp and Levelt [“Tonal Consonance and Critical Bandwidth” R.Plomp, W. J. M. Levelt JASA, Vol 38, 1965], and states that two partialsare considered dissonant if the frequency difference is withinapproximately 5 to 50% of the bandwidth of the critical band in whichthe partials are situated. The scale used for mapping frequency tocritical bands is called the Bark scale. One bark is equivalent to afrequency distance of one critical band. For reference, the function$\begin{matrix}{{z(f)} = {\frac{26.81}{1 + \frac{1960}{f}} - {0.53\quad\lbrack{Bark}\rbrack}}} & (1)\end{matrix}$

[0005] can be used to convert from frequency (f) to the bark scale (z).Plomp states that the human auditory system can not discriminate twopartials if they differ in frequency by approximately less than fivepercent of the critical band in which they are situated, orequivalently, are separated less than 0,05 Bark in frequency. On theother hand, if the distance between the partials are more thanapproximately 0,5 Bark, they will be perceived as separate tones.

[0006] Dissonance theory partly explains why prior-art methods giveunsatisfactory performance. A set of consonant partials translatedupwards in frequency may become dissonant. Moreover, in the interfere,since they may not be within the limits of acceptable deviationaccording to the dissonance-rules.

[0007] WO 98/57436 discloses to perform frequency transposition by meansof multiplication by a transposition factor M. Consecutive channels froman analysis filter bank are frequency-translated to synthesis filterbank channels, but which are spaced apart by two intermediatereconstruction range channels, when the multiplication factor M is 3, orwhich are spaced apart by one reconstruction range channel, when themultiplication factor M equals two. Alternatively, amplitude and phaseinformation from different analyser channels can be combined. Theamplitude signals are connected such that the magnitudes of consecutivechannels of the analysis filterbank are frequency-translated to themagnitudes of subband signals associated with consecutive synthesischannels. The phases of the subband signals from the same channels aresubjected to frequency-transposition using a factor M.

[0008] It is an object of the present invention to provide a concept forobtaining an envelope-adjusted and frequency-translated signal byhigh-frequency spectral reconstruction and a concept for decoding usinghigh-frequency spectral reconstruction, that result in a better qualityreconstruction.

[0009] This object is achieved by a method in accordance with claims 1and 13 and 23 or an apparatus according to claims 19 and 20 or a decoderaccording to claim 21.

[0010] The present invention provides a new method and device forimprovements of translation or folding techniques in source codingsystems. The objective includes substantial reduction of computationalcomplexity and reduction of perceptual artifacts. The invention shows anew implementation of a subsampled digital filter bank as a frequencytranslating or folding device, also offering improved crossover accuracybetween the lowband and the translated or folded bands. Further, theinvention teaches that crossover regions, to avoid sensory dissonance,benefits from being filtered. The filtered regions are called dissonanceguard-bands, and the invention offers the possibility to reducedissonant partials in an uncomplicated and accurate manner using thesubsampled filterbank

[0011] The new filterbank based translation or folding process mayadvantageously be integrated with the spectral envelope adjustmentprocess. The filterbank used for envelope adjustment is then used forthe frequency translation or folding process as well, in that wayeliminating the need to use a separate filterbank or process forspectral envelope adjustment. The proposed invention offers a unique andflexible filterbank design at a low computational cost, thus creating avery effective translation/folding/envelope-adjusting system.

[0012] In addition, the proposed invention is advantageously combinedwith the Adaptive Noise-Floor Addition method described in PCT patent[SE00/00159]. This combination will improve the perceptual quality underdifficult programme material conditions.

[0013] The proposed subband domain based translation of foldingtechnique comprise the following steps:

[0014] filtering of a lowband signal through the analysis part of adigital filterbank to obtain a set of subband signals;

[0015] repatching of a number of the subband signals from consecutivelowband channels to consecutive higbband channels in the synthesis partof a digital fliterbank;

[0016] adjustment of the patched subband signals, in accordance to adesired spectral envelope; and

[0017] filtering of the adjusted subband signals through the synthesispart of a digital filterbank, to obtain an envelope adjusted andfrequency translated or folded signal in a very effective way.

[0018] Attractive applications of the proposed invention relates to theimprovement of various types of intermediate quality codec applications,such as MPEG 2 Layer a, MPEG 2/4 AAC, Dolby AC-3, NTT TwinVQ,AT&T/Lucent PAC etc. where such codecs are used at low bitrates. Theinvention is also very useful in various speech codecs such as G. 729MPEG-4 CELP and HVXC etc to improve perceived quality. The above codersare widely used in multimedia, in the telephone industry, on theInternet as well as in professional multimedia applications.

[0019] The present invention is described by way of illustrativeexamples, not limiting the scope or spirit of the invention, withreference to the accompanying drawings, in which:

[0020]FIG. 1 illustrates filterbank-based translation or foldingintegrated in a coding system according to the present invention;

[0021]FIG. 2 shows a basic structure of a maximally decimatedfilterbank;

[0022]FIG. 3 illustrates spectral translation according to the presentinvention;

[0023]FIG. 4 illustrates spectral folding according to the presentinvention;

[0024]FIG. 5 illustrates spectral translation using guard-bandsaccording to the present invention.

DIGITAL FILTERBANK BASED TRANSLATION AND FOLDING

[0025] New filter bank based translating or folding techniques will nowbe described The signal under consideration is decomposed into a seriesof subband signals by the analysis part of the filterbank. The subbandsignals are then repatched, through reconnection of analysis- andsynthesis subband channels, to achieve spectral translation or foldingor a combination thereof.

[0026]FIG. 2 shows the basic structure of a maximally decimatedfilterbank analysis/synthesis system The analysis filter bank 201 splitsthe input signal into several subband signals. The synthesis FIG. 2shows the basic structure of a maximally decimated filterbankanalysis/synthesis system. The analysis filter bank 201 splits the inputsignal into several subband signals. The synthesis filter bank 202combines the subband samples in order to recreate the original signal.Implementations using maximally decimated filter banks will drasticallyreduce computational costs. It should be appreciated, that the inventioncan be implemented using several types of filter banks or transforms,including cosine or complex exponential modulated filter banks, filterbank interpretations of the wavelet transform, other non-equal bandwidthfilter banks or transforms and multi-dimensional filter banks ortransforms.

[0027] In the illustrative, but not limiting, descriptions below it isassumed that an L-channel filter bank splits the input signal x(n) intoL subband signals. The input signal, with sampling frequency ƒs, isbandlimited to frequency ƒc. The analysis filters of a maximallydecimated filter bank (FIG. 2) are denoted H_(k)(z) 203, where k=0, 1, .. . , L-1. The subband signals ν_(k)(n) are maximally decimated, each ofsampling frequency ƒ_(s)/L, after passing the decimators 204, Thesynthesis section, with the synthesis filters denoted F_(k)(z),reassembles the subband signals after interpolation 205 and filtering206 to produce {circumflex over (x)}(n). In addition, the presentinvention performs a spectral reconstruction on {circumflex over(x)}(n), giving an enhanced signal y(n).

[0028] The reconstruction range start channel, denoted M, is determinedby $\begin{matrix}{M = {{floor}\quad {\left\{ {\frac{f_{c}}{f_{s}}2L} \right\}.}}} & (2)\end{matrix}$

[0029] The number of source area channels is denoted S (1≦S ≦M).Performing spectral reconstruction through translation on {circumflexover (x)}(n) according to the present invention, in combination withenvelope adjustment, is accomplished by repatching the subband signalsas

^(ν) M+k ^((n)=e) M+k ^((n)ν) M−S−P+k ^((n))   (3)

[0030] where k ε [0, S−1], (−1)^(S+P)=1, i.e. S+P is an even number, Pis an integer offset (0≦P≦M−S) and e_(M+k)(n) is the envelopecorrection. Performing spectral reconstruction through folding on{circumflex over (x)}(n) according to the present invention, is furtheraccomplished by repatching the subband signals as

^(ν) M+k ^((n)=e) M+k ^((n)ν*) M−P−S−k ^((n))   (4)

[0031] where k ε [0, S−1], (−1)^(S+P)=−1, i.e. S+P is an odd integernumber, P is an integer offset (1−S≦P≦M−2S+1) and e_(M+k)(n) is theenvelope correction. The operator [*] denotes complex conjugation.Usually, the repatching process is repeated until the intended amount ofhigh frequency bandwidth is attained.

[0032] It should be noted that, through the use of the subband domainbased translation and folding, improved crossover accuracy between thelowband and instances of translated or folded bands is achieved, sinceall the signals are filtered through filterbank channels that havematched frequency responses.

[0033] If the frequency ƒ_(c) of x(n) is too high, or equivalently ƒ_(s)is too low, to allow an effective spectral reconstruction, i.e. M+S>L,the number of subband channels may be increased after the analysisfiltering. Filtering the subband signals with a QL-channel synthesisfilter bank, where only the L lowband channels are used and theupsampling factor Q is chosen so that QL is an integer value, willresult in an output signal with sampling frequency Qƒ_(s). Hence, theextended filter bank will act as if it is an L-channel filter bankfollowed by an upsampler. Since, in this case, the L(Q-1) highbandfilters are unused (fed with zeros), the audio bandwidth will notchange—the filter bank will merely reconstruct an upsampled version of{circumflex over (x)}(n). If, however, the L subband signals arerepatched to the highband channels, according to Eq.(3) or (4), thebandwidth of x(n) will be increased. Using this scheme, the upsamplingprocess is integrated in the synthesis filtering. It should be notedthat any size of the synthesis filter bank may be used, resulting indifferent sampling rates of the output signal.

[0034] Referring to FIG. 3, consider the subband channels from a16-channel analysis filterbank. The input signal x(n) has frequencycontents up to the Nyqvist frequency (ƒ_(c=ƒ) _(s)/2). In the firstiteration, the 16 subbands are extended to 23 subbands, and frequencytranslation according to Eq.(3) is used with the following parameters:M=16, S=7 and P=1. This operation is illustrated by the repatching ofsubbands from point a to b in the figure. In the next iteration, the 23subbands are extended to 28 subbands, and Eq.(3) is used with the newparameters: M=23, S=5 and P=3. This operation is illustrated by therepatching of subbands from point b to c. The so-produced subbands maythen be synthesized using a 28-channel filterbank. This would produce acritically sampled output signal with sampling frequency 28/16ƒ_(s)=1.75ƒ_(s). The subband signals could also be synthesized using a32-channel filterbank, where the four uppermost channels are fed withzeros, illustrated by the dashed lines in the figure, producing anoutput signal with sampling frequency 2ƒ_(s).

[0035] Using the same analysis filterbank and an input signal with thesame frequency contents, FIG. 4 illustrates the repatching usingfrequency folding according to Eq.(4) in two iterations. In the firstiteration M=16, S=8 and P=−7, and the 16 subbands are extended to 24. Inthe second iteration M=24, S=8 and P=−7, and the number of subbands areextended from 24 to 32. The subbands are synthesized with a 32-channelfilterbank. In the output signal, sampled at frequency ^(2ƒ) _(s), thisrepatching results in two reconstructed frequency bands—one bandemerging from the repatching of subband signals to channels 16 to 23,which is a folded version of the bandpass signal extracted by channels 8to 15, and one band emerging from the repatching to channels 24 to 31,which is a translated version of the same bandpass signal.

[0036] Guardbands in High Frequency Reconstruction

[0037] Sensory dissonance may develop in the translation or foldingprocess due to adjacent band interference, i.e. interference betweenpartials in the vicinity of the crossover region between instances oftranslated bands and the lowband. This type of dissonance is more commonin harmonic rich, multiple pitched programme material. In order toreduce dissonance, guard-bands are inserted and may preferably consistof small frequency bands with zero energy, i.e. the crossover regionbetween the lowband signal and the replicated spectral band is filteredusing a bandstop or notch filter. Less perceptual degradation will beperceived if dissonance reduction using guard-bands is performed. Thebandwidth of the guard-bands should preferably be around 0,5 Bark. Ifless, dissonance may result and if wider, comb-filter-like soundcharacteristics may result.

[0038] In filterbank based translation or folding, guard-bands could beinserted and may preferably consist of one or several subband channelsset to zero. The use of guardbands changes Eq.(3) to

^(ν) M+D+k ^((n)=e) M+D+k ^((n)ν) M−S−P+k ^((n))   (5)

[0039] and Eq.(4) to

^(ν) M+D+k ^((n)=e) M+D+k ^((n)ν*) M−P−S−k ^((n))   (6)

[0040] D is a small integer and represents the number of filterbankchannels used as guardband. Now P+S+D should be an even integer inEq.(5) and an odd integer in Eq.(6). P takes the same values as before.FIG. 5 shows the repatching of a 32-channel filterbank using Eq.(5). Theinput signal has frequency contents up to ƒ_(c)={fraction (5/16)}ƒ_(s),making M=20 in the first iteration. The number of source channels ischosen as S=4 and P=2. Further, D should preferably be chosen as to makethe bandwidth of the guardbands 0,5 Bark. Here, D equals 2, making theguardbands ƒ_(s)/32 Hz wide. In the second iteration, the parameters arechosen as M=26, S=4, D=2 and P=0. In the figure, the guardbands areillustrated by the subbands with the dashed line-connections.

[0041] In order to make the spectral envelope continuous, the dissonanceguard-bands may be partially reconstructed using a random white noisesignal, i.e. the subbands are fed with white noise instead of beingzero. The preferred method uses Adaptive Noise-floor Addition (ANA) asdescribed in the PCT patent application [SE00/00159]. This methodestimates the noise-floor of the highband of the original signal andadds synthetic noise in a well-defined way to the recreated highband inthe decoder.

[0042] Practical Implementations

[0043] The present invention may be implemented in various kinds ofsystems for storage or transmission of audio signals using arbitrarycodecs. FIG. 1 shows the decoder of an audio coding system. Thedemultiplexer 101 separates the envelope data and other HFR relatedcontrol signals from the bitstream and feeds the relevant part to thearbitrary lowband decoder 102. The lowband decoder produces a digitalsignal which is fed to the analysis filterbank 104. The envelope data isdecoded in the envelope decoder 103, and the resulting spectral envelopeinformation is fed together with the subband samples from the analysisfilterbank to the integrated translation or folding and envelopeadjusting filterbank unit 105. This unit translates or folds the lowbandsignal, according to the present invention, to form a wideband signaland applies the transmitted spectral envelope. The processed subbandsamples are then fed to the synthesis filterbank 106, which might be ofa different size than the analysis filterbank. The digital widebandoutput signal is finally converted 107 to an analogue output signal.

[0044] The above-described embodiments are merely illustrative for theprinciples of the present invention for improvement of High FrequencyReconstruction (HER) techniques using filterbank-based frequencytranslation or folding. It is understood that modifications andvariations of the arrangements and the details described herein will beapparent to others skilled in the art. It is the intent, therefore, tobe limited only by the scope of the impending patent claims and not bythe specific details presented by way of description and explanation ofthe embodiments herein.

1. Method for obtaining an envelope adjusted and frequency-translated signal by high-frequency spectral reconstruction of complex subband signals in channels within a reconstruction range using complex subband signals in source area channels derived from a lowband signal, using a digital filter bank having an analysis part (201) and a synthesis part (202), the reconstruction range including channel frequencies which are higher than frequencies in the source area channels, the method comprising the following steps: filtering the lowband signal by means of the analysis part (201) to obtain the complex subband signals in the source area channels; calculating a number of consecutive complex subband signals in channels within the reconstruction range using a number of frequency-translated consecutive complex subband signals in the source area channels and an envelope correction for obtaining a predetermined spectral envelope, wherein a complex subband signal in a source area channel having an index i is frequency-translated to a complex subband signal in a reconstruction range channel having an index j, and wherein a complex subband signal in a source area channel having an index i+1 is frequency-translated to a complex subband signal in a reconstruction range channel having an index j+1, and filtering the consecutive complex subband signals in channels within the reconstruction range by means of the synthesis part to obtain an envelope adjusted and frequency-translated signal.
 2. Method according to claim 1, in which, in the step of calculating, the following equation is used: ^(ν) M+k ^((n)=e) M+k ^((n)ν) M−S−P+k ^((n)), wherein M indicates a number of a channel of the synthesis part (202), the channel being a start channel of the reconstruction range, wherein S indicates the number of source area channels, S being a integer greater than or equal to 1 and lower than or equal to M, wherein P is an integer offset greater than or equal to 0 and lower than or equal to M−S; wherein v_(i) indicates a band pass signal v for a channel i of the synthesis part, wherein e_(i) indicates an envelope correction for a channel i of the synthesis part to obtain the desired spectral envelope, wherein n is a time index, and wherein k is an integer index between zero and S−1.
 3. Method according to claim 2, wherein S and P are selected such that a sum of S and P is an even number.
 4. Method according to one of the preceding claims, wherein the digital filterbank is obtained by cosine or sine modulation of a low pass prototype filter
 5. Method according to one of claims 1 to 3, wherein the digital filterbank is obtained by complex-exponential-modulation of a low pass prototype filter.
 6. Method according to claims 4 or 5, wherein the low pass prototype filter is designed so that a transition band of the channels of the digital filterbank overlaps a pass band of the neighbouring channels only.
 7. Method according to one of the preceding claims, in which the synthesis part includes a dissonance guard band, the dissonance guard band being positioned between the source area channels and the reconstruction range channels.
 8. Method according to claim 7, wherein, in the step of calculation, the following equation is used: ^(ν) M+D+k ^((n)=e) M+D+k ^((n)ν) M−s−P+k ^((n)), wherein D is an integer representing a number of filterbank channels used as the dissonance guard band.
 9. Method according to claim 1, wherein P, S, D are selected such that a sum of P, S and D is an even integer.
 10. Method according to one of claims 7 to 9, in which one or several of the channels in the dissonance guard band are fed with zeros or gaussian noise, whereby dissonance related artefacts are attenuated.
 11. Method according to one of claims 7 to 10, in which a bandwidth of the dissonance guard band is approximately one half Bark.
 12. Method according to one of the preceding claims, in which the step of calculating implements a first iteration step, and in which the method further includes another step of calculating, implementing a second iteration step, wherein in the second iteration step, the source area channels include the reconstruction-arranged channels from the first iteration step.
 13. Method for obtaining an envelope adjusted and frequency-folded signal by high-frequency spectral reconstruction of complex subband signals in channels within a reconstruction range using complex subband signals in source area channels derived from a lowband signal, using a digital filter bank having an analysis part (201) and a synthesis part (202), the reconstruction range including channel frequencies which are higher than frequencies in the source area channels, the method comprising the following steps: filtering the lowband signal by means of the analysis part (201) to obtain the complex subband signals in the source area channels; calculating a number of consecutive complex subband signals in channels within the reconstruction range using a number of frequency-translated consecutive conjugate complex subband signals in the source area channels and an envelope correction for obtaining a predetermined spectral envelope, wherein a complex subband signal in a source area channel having an index i is frequency-folded to a complex subband signal in a reconstruction range channel having an index j, and wherein a complex subband signal in a source area channel having an index i+1 is frequency-folded to a complex subband signal in a reconstruction range channel having an index j−1, and filtering the consecutive complex subband signals in channels within the reconstruction range by means of the synthesis part to obtain an envelope adjusted and frequency-translated signal.
 14. Method according to claim 13, in which, in the step of calculating, the following equation is used: ^(ν) M+k ^((n)=e) M+k ^((n)ν*) M−P−S+k ^((n)), wherein M indicates a number of a channel of the synthesis part (202), the channel being a start channel of the reconstruction range, wherein S indicates the number of source area channels, S being a integer greater than or equal to 1 and lower than or equal to M, wherein P is an integer offset greater than or equal to 1−S and lower than or equal to M−2S+1; wherein v_(i) indicates a band pass signal v for a channel i of the synthesis part, wherein e_(i) indicates an envelope correction for a channel i of the synthesis part to obtain the desired spectral envelope, wherein * indicates conjugate complex, wherein n is a time index, and wherein k is an integer index between zero and S−1.
 15. Method according to claim 14, wherein S and P are selected such that a sum of S and P is an odd integer number.
 16. Method according to claim 13, in which the synthesis part includes a dissonance guard band, the dissonance guard band being positioned between the source area channels and the reconstruction range channels.
 17. Method according to claim 16, wherein, in the step of calculation, the following equation is used: ^(ν) M+D+k ^((n)=e) M+D+k ^((n)ν*) M−P−S−k ^((n)), wherein D is an integer representing a number of filterbank channels used as the dissonance guard band.
 18. Method according to claim 1, wherein P, S, D are selected such that a sum of P, S and D is an odd integer.
 19. Apparatus for obtaining an envelope adjusted and frequency-translated signal by high-frequency spectral reconstruction of complex subband signals in channels within a reconstruction range using complex subband signals in source area channels derived from a lowband signal, using a digital filter bank having an analysis part (201) and a synthesis part (202), the reconstruction range including channel frequencies which are higher than frequencies in the source area channels, comprising: means for filtering the lowband signal by means of the analysis part (201) to obtain the complex subband signals in the source area channels; means for calculating a number of consecutive complex subband signals in channels within the reconstruction range using a number of frequency-translated consecutive complex subband signals in the source area channels and an envelope correction for obtaining a predetermined spectral envelope, wherein a complex subband signal in a source area channel having an index i is frequency-translated to a complex subband signal in a reconstruction range channel having an index j, and wherein a complex subband signal in a source area channel having an index i+1 is frequency-translated to a complex subband signal in a reconstruction range channel having an index j+1, and means for filtering the consecutive complex subband signals in channels within the reconstruction range by means of the synthesis part to obtain an envelope adjusted and frequency-translated signal.
 20. Apparatus for obtaining an envelope adjusted and frequency-folded signal by high-frequency spectral reconstruction of complex subband signals in channels within a reconstruction range using complex subband signals in source area channels derived from a lowband signal, using a digital filter bank having an analysis part (201) and a synthesis part (202), the reconstruction range including channel frequencies which are higher than frequencies in the source area channels, comprising: means for filtering the lowband signal by means of the analysis part (201) to obtain the complex subband signals in the source area channels; means for calculating a number of consecutive complex subband signals in channels within the reconstruction range using a number of frequency-translated consecutive conjugate complex subband signals in the source area channels and an envelope correction for obtaining a predetermined spectral envelope, wherein a complex subband signal in a source area channel having an index i is frequency-folded to a complex subband signal in a reconstruction range channel having an index j, and wherein a complex subband signal in a source area channel having an index i+1 is frequency-folded to a complex subband signal in a reconstruction range channel having an index j−1, and means for filtering the consecutive complex subband signals in channels within the reconstruction range by means of the synthesis part to obtain an envelope adjusted and frequency-translated signal.
 21. Decoder for decoding coded signals, the coded signals including a coded lowband audio signal, comprising: a separator (101) for separating the coded lowband audio signal from the coded signals; an audio decoder (102) for audio decoding the coded lowband audio signal to obtain an audio decoded signal; an apparatus in accordance with claim 19 or claim 20, to obtain an envelope-adjusted and frequency-translated or frequency-folded signal using the audio decoded signal as the lowband signal, wherein the envelope-adjusted and frequency-translated or frequency-coded signal is a high-frequency reconstructed version of the lowband audio signal.
 22. Decoder according to claim 21, in which the coded signals further include envelope data, in which the separator (101) is further arranged to separate the envelope data from the coded signals, wherein the decoder further includes an envelope decoder (103) for decoding the envelope data to obtain spectral envelope information, wherein the spectral envelope information is fed to the apparatus for obtaining an envelope adjusted and frequency-translated or frequency-folded signal to be used as an envelope correction for obtaining the predetermined spectral envelope.
 23. Method for decoding coded signals, the coded signals including a coded lowband audio signal, the method comprising the following steps: separating (101) the coded lowband audio signal from the coded signals; audio decoding (102) the coded lowband audio signal to obtain an audio decoded signal; a method in accordance with claim 1 or claim 13, to obtain an envelope-adjusted and frequency-translated or frequency-folded signal using the audio decoded signal as the lowband signal, wherein the envelope-adjusted and frequency-translated or frequency-coded signal is a high-frequency reconstructed version of the lowband audio signal. 